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dcsctp: Add Data Tracker The Data Tracker's purpose is to keep track of all received DATA chunks and to ACK/NACK that data, by generating SACK chunks reflecting its view of what has been received and what has been lost. It also contains logic for _when_ to send the SACKs, as that's different depending on e.g. packet loss. Generally, SACKs are sent every second packet on a connection with no packet loss, and can also be sent on a delayed timer. In case partial reliability is used, and the transmitter has decided that some data shouldn't be retransmitted, it will send a FORWARD-TSN chunk, which this class also handles, by "forgetting" about those chunks. Bug: webrtc:12614 Change-Id: Ifafb0c211f6a47872e81830165ab5fc43ee7f366 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213664 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33676}
2021-04-01 23:36:03 +02:00
# Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../../webrtc.gni")
rtc_library("data_tracker") {
deps = [
"../../../api:array_view",
"../../../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../../rtc_base:logging",
dcsctp: Add Data Tracker The Data Tracker's purpose is to keep track of all received DATA chunks and to ACK/NACK that data, by generating SACK chunks reflecting its view of what has been received and what has been lost. It also contains logic for _when_ to send the SACKs, as that's different depending on e.g. packet loss. Generally, SACKs are sent every second packet on a connection with no packet loss, and can also be sent on a delayed timer. In case partial reliability is used, and the transmitter has decided that some data shouldn't be retransmitted, it will send a FORWARD-TSN chunk, which this class also handles, by "forgetting" about those chunks. Bug: webrtc:12614 Change-Id: Ifafb0c211f6a47872e81830165ab5fc43ee7f366 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213664 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33676}
2021-04-01 23:36:03 +02:00
"../../../rtc_base:rtc_base_approved",
Remove //rtc_base:stringutils from public deps Bug: webrtc:8603 Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36453}
2022-04-04 17:14:02 +02:00
"../../../rtc_base:stringutils",
dcsctp: Add missing target dependencies Those were found when trying to build within Chromium's codebase. Bug: webrtc:12614 Change-Id: Ic3f7a266ad4b5d816a693645e1e909fc39d513c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217220 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33896}
2021-05-03 14:07:28 +02:00
"../common:sequence_numbers",
"../packet:chunk",
"../packet:data",
dcsctp: implement handover in DataTracker Bug: webrtc:13154 Change-Id: Ia8c41dcffd95dafd904ee630f2131b575fe833dd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231955 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34996}
2021-09-14 22:08:53 +02:00
"../public:socket",
dcsctp: Add missing target dependencies Those were found when trying to build within Chromium's codebase. Bug: webrtc:12614 Change-Id: Ic3f7a266ad4b5d816a693645e1e909fc39d513c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217220 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33896}
2021-05-03 14:07:28 +02:00
"../timer",
dcsctp: Add Data Tracker The Data Tracker's purpose is to keep track of all received DATA chunks and to ACK/NACK that data, by generating SACK chunks reflecting its view of what has been received and what has been lost. It also contains logic for _when_ to send the SACKs, as that's different depending on e.g. packet loss. Generally, SACKs are sent every second packet on a connection with no packet loss, and can also be sent on a delayed timer. In case partial reliability is used, and the transmitter has decided that some data shouldn't be retransmitted, it will send a FORWARD-TSN chunk, which this class also handles, by "forgetting" about those chunks. Bug: webrtc:12614 Change-Id: Ifafb0c211f6a47872e81830165ab5fc43ee7f366 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213664 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33676}
2021-04-01 23:36:03 +02:00
]
sources = [
"data_tracker.cc",
"data_tracker.h",
]
Enable GN check on //net. This should avoid the situation where WebRTC's GN check is green and Chromium (which turns it ON for //third_party/webrtc) fails. Bug: webrtc:12614 Change-Id: Id4c06ac57e9faa07c5e43491a61fbc093c68a40d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33900}
2021-05-03 14:39:25 +02:00
absl_deps = [
dcsctp: Optimize SACK generation Before this CL, a SACK was generated from scratch based on information about each received fragment, to generate correct gap-ack-blocks. When there was a lot of data in the data tracker (due to packet loss), this took considerate time, as generating a SACK was O(N), where N is the amount of fragments in the data tracker. By instead having precomputed gap-ack-blocks that are continuously updated, generating a SACK is much faster and the memory usage goes down a bit as well. Bug: webrtc:12799 Change-Id: I924752c1d6d31f06d27246e10b595e9ccb19320f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220763 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34171}
2021-05-29 21:20:12 +02:00
"//third_party/abseil-cpp/absl/algorithm:container",
Enable GN check on //net. This should avoid the situation where WebRTC's GN check is green and Chromium (which turns it ON for //third_party/webrtc) fails. Bug: webrtc:12614 Change-Id: Id4c06ac57e9faa07c5e43491a61fbc093c68a40d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33900}
2021-05-03 14:39:25 +02:00
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
dcsctp: Add Data Tracker The Data Tracker's purpose is to keep track of all received DATA chunks and to ACK/NACK that data, by generating SACK chunks reflecting its view of what has been received and what has been lost. It also contains logic for _when_ to send the SACKs, as that's different depending on e.g. packet loss. Generally, SACKs are sent every second packet on a connection with no packet loss, and can also be sent on a delayed timer. In case partial reliability is used, and the transmitter has decided that some data shouldn't be retransmitted, it will send a FORWARD-TSN chunk, which this class also handles, by "forgetting" about those chunks. Bug: webrtc:12614 Change-Id: Ifafb0c211f6a47872e81830165ab5fc43ee7f366 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213664 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33676}
2021-04-01 23:36:03 +02:00
}
dcsctp: Add Traditional Reassembly Streams This class handles the assembly of fragmented received messages (as DATA chunks) and manage per-stream queues. This class only handles non-interleaved messages as described in RFC4960, and is not used when message interleaving is enabled on the association, as described in RFC8260. This is also only part of the reassembly - a follow-up change will add the ReassemblyQueue that handle the other part as well. And an even further follow-up change will add a "interleaved reassembly stream". Bug: webrtc:12614 Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33677}
2021-04-03 19:39:46 +02:00
rtc_source_set("reassembly_streams") {
dcsctp: Add missing target dependencies Those were found when trying to build within Chromium's codebase. Bug: webrtc:12614 Change-Id: Ic3f7a266ad4b5d816a693645e1e909fc39d513c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217220 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33896}
2021-05-03 14:07:28 +02:00
deps = [
"../../../api:array_view",
"../common:sequence_numbers",
"../packet:chunk",
"../packet:data",
dcsctp: introduce handover API types and implement it for streams Bug: webrtc:13154 Change-Id: Ifa250175af79b7adc87dbc2750054adc94b90bb7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231842 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34991}
2021-09-14 13:59:55 +02:00
"../public:socket",
dcsctp: Add missing target dependencies Those were found when trying to build within Chromium's codebase. Bug: webrtc:12614 Change-Id: Ic3f7a266ad4b5d816a693645e1e909fc39d513c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217220 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33896}
2021-05-03 14:07:28 +02:00
"../public:types",
]
dcsctp: Add Traditional Reassembly Streams This class handles the assembly of fragmented received messages (as DATA chunks) and manage per-stream queues. This class only handles non-interleaved messages as described in RFC4960, and is not used when message interleaving is enabled on the association, as described in RFC8260. This is also only part of the reassembly - a follow-up change will add the ReassemblyQueue that handle the other part as well. And an even further follow-up change will add a "interleaved reassembly stream". Bug: webrtc:12614 Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33677}
2021-04-03 19:39:46 +02:00
sources = [ "reassembly_streams.h" ]
dcsctp: Add missing target dependencies Those were found when trying to build within Chromium's codebase. Bug: webrtc:12614 Change-Id: Ic3f7a266ad4b5d816a693645e1e909fc39d513c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217220 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33896}
2021-05-03 14:07:28 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
dcsctp: Add Traditional Reassembly Streams This class handles the assembly of fragmented received messages (as DATA chunks) and manage per-stream queues. This class only handles non-interleaved messages as described in RFC4960, and is not used when message interleaving is enabled on the association, as described in RFC8260. This is also only part of the reassembly - a follow-up change will add the ReassemblyQueue that handle the other part as well. And an even further follow-up change will add a "interleaved reassembly stream". Bug: webrtc:12614 Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33677}
2021-04-03 19:39:46 +02:00
}
rtc_library("traditional_reassembly_streams") {
deps = [
":reassembly_streams",
"../../../api:array_view",
"../../../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../../rtc_base:logging",
dcsctp: Add Traditional Reassembly Streams This class handles the assembly of fragmented received messages (as DATA chunks) and manage per-stream queues. This class only handles non-interleaved messages as described in RFC4960, and is not used when message interleaving is enabled on the association, as described in RFC8260. This is also only part of the reassembly - a follow-up change will add the ReassemblyQueue that handle the other part as well. And an even further follow-up change will add a "interleaved reassembly stream". Bug: webrtc:12614 Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33677}
2021-04-03 19:39:46 +02:00
"../../../rtc_base:rtc_base_approved",
dcsctp: Add missing target dependencies Those were found when trying to build within Chromium's codebase. Bug: webrtc:12614 Change-Id: Ic3f7a266ad4b5d816a693645e1e909fc39d513c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217220 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33896}
2021-05-03 14:07:28 +02:00
"../common:sequence_numbers",
"../packet:chunk",
"../packet:data",
"../public:types",
dcsctp: Add Traditional Reassembly Streams This class handles the assembly of fragmented received messages (as DATA chunks) and manage per-stream queues. This class only handles non-interleaved messages as described in RFC4960, and is not used when message interleaving is enabled on the association, as described in RFC8260. This is also only part of the reassembly - a follow-up change will add the ReassemblyQueue that handle the other part as well. And an even further follow-up change will add a "interleaved reassembly stream". Bug: webrtc:12614 Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33677}
2021-04-03 19:39:46 +02:00
]
sources = [
"traditional_reassembly_streams.cc",
"traditional_reassembly_streams.h",
]
dcsctp: Add missing target dependencies Those were found when trying to build within Chromium's codebase. Bug: webrtc:12614 Change-Id: Ic3f7a266ad4b5d816a693645e1e909fc39d513c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217220 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33896}
2021-05-03 14:07:28 +02:00
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
dcsctp: Add Traditional Reassembly Streams This class handles the assembly of fragmented received messages (as DATA chunks) and manage per-stream queues. This class only handles non-interleaved messages as described in RFC4960, and is not used when message interleaving is enabled on the association, as described in RFC8260. This is also only part of the reassembly - a follow-up change will add the ReassemblyQueue that handle the other part as well. And an even further follow-up change will add a "interleaved reassembly stream". Bug: webrtc:12614 Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33677}
2021-04-03 19:39:46 +02:00
}
dcsctp: Add Reassembly Queue The Reassembly Queue receives fragmented messages (DATA or I-DATA chunks) and - with help of stream reassemblers - will reassemble these fragments into messages, which will be delivered to the client. It also handle partial reliability (FORWARD-TSN) and stream resetting. To avoid a DoS attack vector, where a sender can send fragments in a way that the reassembly queue will never succeed to reassemble a message and use all available memory, the ReassemblyQueue has a maximum size. Bug: webrtc:12614 Change-Id: Ibb084fecd240d4c414e096579244f8f5ee46914e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214043 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33678}
2021-04-03 20:33:43 +02:00
rtc_library("reassembly_queue") {
deps = [
dcsctp: Add missing target dependencies Those were found when trying to build within Chromium's codebase. Bug: webrtc:12614 Change-Id: Ic3f7a266ad4b5d816a693645e1e909fc39d513c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217220 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33896}
2021-05-03 14:07:28 +02:00
":reassembly_streams",
dcsctp: Add Reassembly Queue The Reassembly Queue receives fragmented messages (DATA or I-DATA chunks) and - with help of stream reassemblers - will reassemble these fragments into messages, which will be delivered to the client. It also handle partial reliability (FORWARD-TSN) and stream resetting. To avoid a DoS attack vector, where a sender can send fragments in a way that the reassembly queue will never succeed to reassemble a message and use all available memory, the ReassemblyQueue has a maximum size. Bug: webrtc:12614 Change-Id: Ibb084fecd240d4c414e096579244f8f5ee46914e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214043 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33678}
2021-04-03 20:33:43 +02:00
":traditional_reassembly_streams",
"../../../api:array_view",
"../../../rtc_base:checks",
Remove //rtc_base:logging from public deps Bug: webrtc:8603 Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-04 15:06:30 +02:00
"../../../rtc_base:logging",
dcsctp: Add Reassembly Queue The Reassembly Queue receives fragmented messages (DATA or I-DATA chunks) and - with help of stream reassemblers - will reassemble these fragments into messages, which will be delivered to the client. It also handle partial reliability (FORWARD-TSN) and stream resetting. To avoid a DoS attack vector, where a sender can send fragments in a way that the reassembly queue will never succeed to reassemble a message and use all available memory, the ReassemblyQueue has a maximum size. Bug: webrtc:12614 Change-Id: Ibb084fecd240d4c414e096579244f8f5ee46914e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214043 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33678}
2021-04-03 20:33:43 +02:00
"../../../rtc_base:rtc_base_approved",
dcsctp: Add missing target dependencies Those were found when trying to build within Chromium's codebase. Bug: webrtc:12614 Change-Id: Ic3f7a266ad4b5d816a693645e1e909fc39d513c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217220 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33896}
2021-05-03 14:07:28 +02:00
"../common:internal_types",
"../common:sequence_numbers",
"../common:str_join",
"../packet:chunk",
"../packet:data",
"../packet:parameter",
dcsctp: support socket handover in ReassemblyQueue Bug: webrtc:13154 Change-Id: I816e51dcd923ba6440480de5d5df9012e4af9e5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231958 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35002}
2021-09-15 11:59:17 +02:00
"../public:socket",
dcsctp: Add missing target dependencies Those were found when trying to build within Chromium's codebase. Bug: webrtc:12614 Change-Id: Ic3f7a266ad4b5d816a693645e1e909fc39d513c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217220 Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33896}
2021-05-03 14:07:28 +02:00
"../public:types",
dcsctp: Add Reassembly Queue The Reassembly Queue receives fragmented messages (DATA or I-DATA chunks) and - with help of stream reassemblers - will reassemble these fragments into messages, which will be delivered to the client. It also handle partial reliability (FORWARD-TSN) and stream resetting. To avoid a DoS attack vector, where a sender can send fragments in a way that the reassembly queue will never succeed to reassemble a message and use all available memory, the ReassemblyQueue has a maximum size. Bug: webrtc:12614 Change-Id: Ibb084fecd240d4c414e096579244f8f5ee46914e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214043 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33678}
2021-04-03 20:33:43 +02:00
]
sources = [
"reassembly_queue.cc",
"reassembly_queue.h",
]
Enable GN check on //net. This should avoid the situation where WebRTC's GN check is green and Chromium (which turns it ON for //third_party/webrtc) fails. Bug: webrtc:12614 Change-Id: Id4c06ac57e9faa07c5e43491a61fbc093c68a40d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33900}
2021-05-03 14:39:25 +02:00
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
dcsctp: Add Reassembly Queue The Reassembly Queue receives fragmented messages (DATA or I-DATA chunks) and - with help of stream reassemblers - will reassemble these fragments into messages, which will be delivered to the client. It also handle partial reliability (FORWARD-TSN) and stream resetting. To avoid a DoS attack vector, where a sender can send fragments in a way that the reassembly queue will never succeed to reassemble a message and use all available memory, the ReassemblyQueue has a maximum size. Bug: webrtc:12614 Change-Id: Ibb084fecd240d4c414e096579244f8f5ee46914e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214043 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33678}
2021-04-03 20:33:43 +02:00
}
dcsctp: Add Data Tracker The Data Tracker's purpose is to keep track of all received DATA chunks and to ACK/NACK that data, by generating SACK chunks reflecting its view of what has been received and what has been lost. It also contains logic for _when_ to send the SACKs, as that's different depending on e.g. packet loss. Generally, SACKs are sent every second packet on a connection with no packet loss, and can also be sent on a delayed timer. In case partial reliability is used, and the transmitter has decided that some data shouldn't be retransmitted, it will send a FORWARD-TSN chunk, which this class also handles, by "forgetting" about those chunks. Bug: webrtc:12614 Change-Id: Ifafb0c211f6a47872e81830165ab5fc43ee7f366 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213664 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33676}
2021-04-01 23:36:03 +02:00
if (rtc_include_tests) {
rtc_library("dcsctp_rx_unittests") {
testonly = true
deps = [
":data_tracker",
dcsctp: Add Reassembly Queue The Reassembly Queue receives fragmented messages (DATA or I-DATA chunks) and - with help of stream reassemblers - will reassemble these fragments into messages, which will be delivered to the client. It also handle partial reliability (FORWARD-TSN) and stream resetting. To avoid a DoS attack vector, where a sender can send fragments in a way that the reassembly queue will never succeed to reassemble a message and use all available memory, the ReassemblyQueue has a maximum size. Bug: webrtc:12614 Change-Id: Ibb084fecd240d4c414e096579244f8f5ee46914e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214043 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33678}
2021-04-03 20:33:43 +02:00
":reassembly_queue",
Enable GN check on //net. This should avoid the situation where WebRTC's GN check is green and Chromium (which turns it ON for //third_party/webrtc) fails. Bug: webrtc:12614 Change-Id: Id4c06ac57e9faa07c5e43491a61fbc093c68a40d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33900}
2021-05-03 14:39:25 +02:00
":reassembly_streams",
dcsctp: Add Traditional Reassembly Streams This class handles the assembly of fragmented received messages (as DATA chunks) and manage per-stream queues. This class only handles non-interleaved messages as described in RFC4960, and is not used when message interleaving is enabled on the association, as described in RFC8260. This is also only part of the reassembly - a follow-up change will add the ReassemblyQueue that handle the other part as well. And an even further follow-up change will add a "interleaved reassembly stream". Bug: webrtc:12614 Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33677}
2021-04-03 19:39:46 +02:00
":traditional_reassembly_streams",
dcsctp: Add Data Tracker The Data Tracker's purpose is to keep track of all received DATA chunks and to ACK/NACK that data, by generating SACK chunks reflecting its view of what has been received and what has been lost. It also contains logic for _when_ to send the SACKs, as that's different depending on e.g. packet loss. Generally, SACKs are sent every second packet on a connection with no packet loss, and can also be sent on a delayed timer. In case partial reliability is used, and the transmitter has decided that some data shouldn't be retransmitted, it will send a FORWARD-TSN chunk, which this class also handles, by "forgetting" about those chunks. Bug: webrtc:12614 Change-Id: Ifafb0c211f6a47872e81830165ab5fc43ee7f366 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213664 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33676}
2021-04-01 23:36:03 +02:00
"../../../api:array_view",
Allow specifying delayed task precision of dcsctp::Timer. Context: The timer precision of PostDelayedTask() is about to be lowered to include up to 17 ms leeway. In order not to break use cases that require high precision timers, PostDelayedHighPrecisionTask() will continue to have the same precision that PostDelayedTask() has today. webrtc::TaskQueueBase has an enum (kLow, kHigh) to decide which precision to use when calling PostDelayedTaskWithPrecision(). See go/postdelayedtask-precision-in-webrtc for motivation and a table of delayed task use cases in WebRTC that are "high" or "low" precision. Most timers in DCSCTP are believed to only be needing low precision (see table), but the delayed_ack_timer_ of DataTracker[1] is an example of a use case that is likely to break if the timer precision is lowered (if ACK is sent too late, retransmissions may occur). So this is considered a high precision use case. This CL makes it possible to specify the precision of dcsctp::Timer. In a follow-up CL we will update delayed_ack_timer_ to kHigh precision. [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/net/dcsctp/rx/data_tracker.cc;l=340 Bug: webrtc:13604 Change-Id: I8eec5ce37044096978b5dd1985fbb00bc0d8fb7e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249081 Reviewed-by: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35809}
2022-01-26 18:38:13 +01:00
"../../../api/task_queue:task_queue",
dcsctp: Add Data Tracker The Data Tracker's purpose is to keep track of all received DATA chunks and to ACK/NACK that data, by generating SACK chunks reflecting its view of what has been received and what has been lost. It also contains logic for _when_ to send the SACKs, as that's different depending on e.g. packet loss. Generally, SACKs are sent every second packet on a connection with no packet loss, and can also be sent on a delayed timer. In case partial reliability is used, and the transmitter has decided that some data shouldn't be retransmitted, it will send a FORWARD-TSN chunk, which this class also handles, by "forgetting" about those chunks. Bug: webrtc:12614 Change-Id: Ifafb0c211f6a47872e81830165ab5fc43ee7f366 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213664 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33676}
2021-04-01 23:36:03 +02:00
"../../../rtc_base:checks",
"../../../rtc_base:gunit_helpers",
"../../../rtc_base:rtc_base_approved",
"../../../test:test_support",
dcsctp: support handover state serialization testing dcSCTP library users can set their custom g_handover_state_transformer_for_test that can serialize and deserialize the state. All dcSCTP handover tests call g_handover_state_transformer_for_test. If some part of the state is serialized incorrectly or is forgotten, high chance that it will fail the tests. Bug: webrtc:13154 Change-Id: I251a099be04dda7611e9df868d36e3a76dc7d1e1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232325 Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35035}
2021-09-20 11:35:59 +02:00
"../common:handover_testing",
Enable GN check on //net. This should avoid the situation where WebRTC's GN check is green and Chromium (which turns it ON for //third_party/webrtc) fails. Bug: webrtc:12614 Change-Id: Id4c06ac57e9faa07c5e43491a61fbc093c68a40d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33900}
2021-05-03 14:39:25 +02:00
"../common:sequence_numbers",
"../packet:chunk",
"../packet:data",
"../public:types",
dcsctp: Add Traditional Reassembly Streams This class handles the assembly of fragmented received messages (as DATA chunks) and manage per-stream queues. This class only handles non-interleaved messages as described in RFC4960, and is not used when message interleaving is enabled on the association, as described in RFC8260. This is also only part of the reassembly - a follow-up change will add the ReassemblyQueue that handle the other part as well. And an even further follow-up change will add a "interleaved reassembly stream". Bug: webrtc:12614 Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33677}
2021-04-03 19:39:46 +02:00
"../testing:data_generator",
Enable GN check on //net. This should avoid the situation where WebRTC's GN check is green and Chromium (which turns it ON for //third_party/webrtc) fails. Bug: webrtc:12614 Change-Id: Id4c06ac57e9faa07c5e43491a61fbc093c68a40d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33900}
2021-05-03 14:39:25 +02:00
"../timer",
dcsctp: Add Traditional Reassembly Streams This class handles the assembly of fragmented received messages (as DATA chunks) and manage per-stream queues. This class only handles non-interleaved messages as described in RFC4960, and is not used when message interleaving is enabled on the association, as described in RFC8260. This is also only part of the reassembly - a follow-up change will add the ReassemblyQueue that handle the other part as well. And an even further follow-up change will add a "interleaved reassembly stream". Bug: webrtc:12614 Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33677}
2021-04-03 19:39:46 +02:00
]
Enable GN check on //net. This should avoid the situation where WebRTC's GN check is green and Chromium (which turns it ON for //third_party/webrtc) fails. Bug: webrtc:12614 Change-Id: Id4c06ac57e9faa07c5e43491a61fbc093c68a40d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33900}
2021-05-03 14:39:25 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
dcsctp: Add Traditional Reassembly Streams This class handles the assembly of fragmented received messages (as DATA chunks) and manage per-stream queues. This class only handles non-interleaved messages as described in RFC4960, and is not used when message interleaving is enabled on the association, as described in RFC8260. This is also only part of the reassembly - a follow-up change will add the ReassemblyQueue that handle the other part as well. And an even further follow-up change will add a "interleaved reassembly stream". Bug: webrtc:12614 Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33677}
2021-04-03 19:39:46 +02:00
sources = [
"data_tracker_test.cc",
dcsctp: Add Reassembly Queue The Reassembly Queue receives fragmented messages (DATA or I-DATA chunks) and - with help of stream reassemblers - will reassemble these fragments into messages, which will be delivered to the client. It also handle partial reliability (FORWARD-TSN) and stream resetting. To avoid a DoS attack vector, where a sender can send fragments in a way that the reassembly queue will never succeed to reassemble a message and use all available memory, the ReassemblyQueue has a maximum size. Bug: webrtc:12614 Change-Id: Ibb084fecd240d4c414e096579244f8f5ee46914e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214043 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33678}
2021-04-03 20:33:43 +02:00
"reassembly_queue_test.cc",
dcsctp: Add Traditional Reassembly Streams This class handles the assembly of fragmented received messages (as DATA chunks) and manage per-stream queues. This class only handles non-interleaved messages as described in RFC4960, and is not used when message interleaving is enabled on the association, as described in RFC8260. This is also only part of the reassembly - a follow-up change will add the ReassemblyQueue that handle the other part as well. And an even further follow-up change will add a "interleaved reassembly stream". Bug: webrtc:12614 Change-Id: Iaf339fa215a2b14926f5cb74f15528392e273f99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214042 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33677}
2021-04-03 19:39:46 +02:00
"traditional_reassembly_streams_test.cc",
dcsctp: Add Data Tracker The Data Tracker's purpose is to keep track of all received DATA chunks and to ACK/NACK that data, by generating SACK chunks reflecting its view of what has been received and what has been lost. It also contains logic for _when_ to send the SACKs, as that's different depending on e.g. packet loss. Generally, SACKs are sent every second packet on a connection with no packet loss, and can also be sent on a delayed timer. In case partial reliability is used, and the transmitter has decided that some data shouldn't be retransmitted, it will send a FORWARD-TSN chunk, which this class also handles, by "forgetting" about those chunks. Bug: webrtc:12614 Change-Id: Ifafb0c211f6a47872e81830165ab5fc43ee7f366 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213664 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33676}
2021-04-01 23:36:03 +02:00
]
}
}
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